Index: webrtc/modules/audio_processing/audio_processing_impl.h |
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h |
index 6f237ae4dc84df14917777fdefd6041106875d65..bbd17191585037e9fa739a1390f5013023a4ba8e 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.h |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h |
@@ -134,6 +134,7 @@ class AudioProcessingImpl : public AudioProcessing { |
int StartDebugRecording(FILE* handle) override; |
int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override; |
int StopDebugRecording() override; |
+ void UpdateHistogramsOnCallEnd() override; |
EchoCancellation* echo_cancellation() const override; |
EchoControlMobile* echo_control_mobile() const override; |
GainControl* gain_control() const override; |
@@ -210,6 +211,8 @@ class AudioProcessingImpl : public AudioProcessing { |
bool was_stream_delay_set_; |
int last_stream_delay_ms_; |
int last_aec_system_delay_ms_; |
+ int stream_delay_jumps_; |
+ int aec_system_delay_jumps_; |
bool output_will_be_muted_ GUARDED_BY(crit_); |